Audio signal processing

ABSTRACT

A method for processing and transducing audio signals. An audio system has a first audio signal and a second audio signal that have amplitudes. A method for processing the audio signals includes dividing the first audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first scaling factor proportional to the amplitude of the second audio signal; and scaling the first spectral band signal by a second scaling factor to create a second signal portion. Other portions of the disclosure include application of the signal processing method to multichannel audio systems, and to audio systems having different combinations of directional loudspeakers, full range loudspeakers, and limited range loudspeakers.

CROSS-REFERENCE TO RELATED APPLICATIONS

[0001] Not applicable.

STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT

[0002] Not applicable.

[0003] The invention relates to audio signal processing in audio systemshaving multiple directional channels, such as so-called “surroundsystems,” and more particularly to audio signal processing that canadapt multiple directional channel systems to audio systems having feweror more loudspeaker locations than the number of directional channels.

BACKGROUND OF THE INVENTION

[0004] For background, reference is made to surround sound systems andU.S. Pat. Nos. 5,809,153 and 5,870,484. It is an important object of theinvention to provide an improved audio signal processing system for theprocessing of directional channels in a multi-channel audio system.

BRIEF SUMMARY OF THE INVENTION

[0005] According to the invention, an audio system has a first audiosignal and a second audio signal having amplitudes. A method forprocessing the audio signals includes dividing the first audio signalinto a first spectral band signal and a second spectral band signal;scaling the first spectral band signal by a first scaling factor tocreate a first signal portion, wherein the first scaling factor isproportional to the amplitude of the second audio signal; and scalingthe first spectral band signal by a second scaling factor to create asecond signal portion.

[0006] In another aspect of the invention. An audio system has a firstaudio signal, a second audio signal and a directional loudspeaker unit.A method for processing the audio signals includes electroacousticallydirectionally transducing the first audio signal to produce a firstsignal radiation pattern; electroacoustically directionally transducingthe second audio signal to produce a second signal radiation pattern,wherein the first signal radiation pattern and the second signalradiation pattern are alternatively and user selectively similar ordifferent.

[0007] In another aspect of the invention, An audio system has a firstaudio signal, a second audio signal, and a third audio signal that issubstantially limited to a frequency range having a lower limit at afrequency that has a corresponding wavelength that approximates thedimensions of a human head. The audio system further includes adirectional loudspeaker unit, and a loudspeaker unit, distinct from thedirectional loudspeaker unit. A method for processing the audio signals,includes electroacoustically directionally transducing by thedirectional loudspeaker unit the first audio signal to produced a firstradiation pattern; electroacoustically directionally transducing by thedirectional loudspeaker unit the second audio signal to produce a secondradiation pattern; and electroacoustically transducing by the distinctloudspeaker unit the third audio signal.

[0008] In another aspect of the invention, an audio system has aplurality of directional channels. A method for processing audio signalsrespectively corresponding to each of the plurality of channels includesdividing a first audio signal into a first audio signal first spectralband signal and a first audio signal second spectral band signal;scaling the first audio signal first spectral band signal by a firstscaling factor to create a first audio signal first spectral band firstportion signal; scaling the first spectral band signal by a secondscaling factor to create a first audio signal first spectral band secondportion signal; dividing a second audio signal into a second audiosignal first spectral band signal and a second audio signal secondspectral band signal; scaling the second audio signal first spectralband signal by a third scaling factor to create a second audio signalfirst spectral band first portion signal; and scaling the second audiosignal first spectral band signal by a fourth scaling factor to create asecond audio signal first spectral band second portion signal.

[0009] In another aspect of the invention, a method for processing anaudio signal includes filtering the signal by a first filter that has afrequency response and time delay effect similar to the human head toproduce a once filtered signal. The method further includes filteringthe once filtered audio signal by a second filter, the second filterhaving a frequency response and time delay effect inverse to thefrequency and time delay effect of a human head on a sound wave.

[0010] In another aspect of the invention, an audio system has aplurality of directional channels, a first audio signal and a secondaudio signal, the first and second audio signals representing adjacentdirectional channels on the same lateral side of a listener in a normallistening position. A method for processing the audio signals includesdividing the first audio signal into a first spectral band signal and asecond spectral band signal; scaling the first spectral band signal by afirst time varying calculated scaling factor to create a first signalportion; and scaling the first spectral band signal by a second timevarying calculated scaling factor to create a second signal portion.

[0011] In still another aspect of the invention, an audio system has anaudio signal, a first electroacoustical transducer designed andconstructed to transduce sound waves in a frequency range having a lowerlimit, and a second electroacoustical transducer designed andconstructed to transduce sound waves in a frequency range having asecond transducer lower limit that is lower than the first transducerlower limit. A method for processing audio signals, includes dividingthe audio signal into a first spectral band signal and a second spectralband signal; scaling the first spectral band signal by a first scalingfactor to create a first portion signal; scaling the first spectral bandsignal by a second scaling factor to create a second portion signal;transmitting the first portion to the first electroacoustical transducerfor transduction; and transmitting said second portion signal to saidsecond electroacoustical transducer for transduction.

[0012] Other features, objects, and advantages will become apparent fromthe following detailed description, which refers to the followingdrawing in which:

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING

[0013]FIGS. 1a-1 c are diagrammatic views of configurations ofloudspeaker units for use with the invention;

[0014]FIG. 2a is a block diagram of an audio signal processing systemincorporating the invention;

[0015]FIGS. 2b and 2 c are block diagrams of audio signal processingsystems FIGS. 1a-1 c are diagrammatic views of configurations ofloudspeaker units for use with the invention;

[0016]FIG. 2a is a block diagram of an audio signal processing systemincorporating the invention;

[0017]FIGS. 2b and 2 c are block diagrams of audio signal processingsystems for creating directional channels in accordance with theinvention;

[0018]FIGS. 3a-3 d are block diagrams of alternate directionalprocessors for use in the audio signal processing system of FIG. 2a;

[0019]FIG. 4 is a block diagram of some of the components of thedirectional processors of FIGS. 3a-3 c;

[0020]FIG. 5 is a diagrammatic view of a configuration of loudspeakershelpful in explaining aspects of the invention;

[0021]FIG. 6 is of a configuration of loudspeaker units for use withanother aspect of the invention;

[0022]FIG. 7 is a block diagram of an audio signal processing systemincorporating another aspect of the invention;

[0023]FIG. 8 is a block diagram of a directional processor for use withthe audio signal processing system of FIG. 7;

[0024]FIG. 9 is a block diagram of an alternate directional processorfor use with the audio signal processing system of FIG. 7;

[0025]FIGS. 10a-10 c are top diagrammatic views of some of thecomponents of an audio system for describing another feature of theinvention; and

[0026]FIG. 11 is a block diagram of a component of FIGS. 3a-3 d.forcreating directional channels in accordance with the invention;

DETAILED DESCRIPTION

[0027] With reference now to the drawing and more particularly to FIGS.1a-1 c, there are shown top diagrammatic views of three configurationsof surround sound audio loudspeaker units according to the invention. InFIG. 1a, two directional arrays each including two full range (asdefined below in the discussion of FIGS. 2a-2 c) acoustical drivers arepositioned in front of a listener 14. A first array 10 includingacoustical drivers 11 and 12 may be positioned to the listener's leftand a second array 15, including acoustical drivers 16 and 17 may bepositioned to the listener's right. In FIG. 1b, two directional arrayseach including two full range acoustical drivers are positioned in frontof a listener 14. A first array 10 including acoustical drivers 11 and12 may be positioned to the listener's left and a second array 15,including acoustical drivers 16 and 17 may be positioned to thelistener's right. In addition, a first limited range (as defined belowin the discussion of FIGS. 2a-2 c) acoustical driver 22 is positionedbehind the listener, to the listener's left, and a second limited rangeacoustical driver 24 is positioned behind the listener to the listener'sright. In FIG. 1c, two directional arrays each including two full rangeacoustical drivers are positioned in front of a listener 14. A firstarray 10 including acoustical drivers 11 and 12 may be positioned to thelistener's left and a second array 15, including acoustical drivers 16and 17 may be positioned to the to the listener's right. In addition, afirst full range acoustical driver 28 is positioned behind the listener,to the listener's left, and a second limited range acoustical driver 30is positioned behind the listener to the listener's right. Othersurround sound loudspeaker systems may have loudspeaker units inadditional locations, such as directly in front of listener 14. Surroundsound systems may radiate sound waves in a manner that the source of thesound may be perceived by the listener to be in a direction (for exampledirection X) relative to the listener at which there is no loudspeakerunit. Surround sound systems may further attempt to radiate sound wavesin a manner such that the source of the sound may be perceived by thelistener to be moving (for example in direction Y-Y′) relative to theviewer

[0028] Referring to FIG. 2a, there is shown a block diagram of an audiosignal processing system for providing audio signals for the loudspeakerunits of FIGS. 1a-1 c. An audio signal source 32 is coupled to a decoder34 which decodes the audio source from the audio signal source into aplurality of channels, in this case a low frequency effects (LFE)channel, and bass channel, and a number of directional channels,including a left surround (LS) channel, a left (L) channel, a leftcenter (LC) channel, a right center (RC) channel, a right (R) channel,and a right surround (RS) channel. Other decoding systems may output adifferent set of channels. In some systems, the bass channel is notbroken out separately from the directional channels, but instead remainscombined with the directional channels. In other systems, there may be asingle center (C) channel, instead of the RC and LC channels, or theremay be a single surround channel. An audio system according to theinvention may be used with any combination of directional channels,either by adapting the signal processing to the channels, or by decodingthe directional channels to produce additional directional channels. Onemethod of decoding a single C channel into an RC channel and an LCchannel is shown in FIG. 2b. The C channel is split into an LC channeland an RC channel and the LC and the RC channel are scaled by a factor,such as 0.707. Similarly, a method of decoding a single S channel intoan RS channel and an LS channel is shown in FIG. 2c. The S channel issplit into an RS channel and an LS channel, and the RS channel and LSchannel are scaled by a factor, such as 0.707. If the audio input signalhas no surround channel or channels, there are several known methods forsynthesizing surround channels from existing channels, or the system maybe operated without surround sound.

[0029] Some surround sound systems have a separate low frequency unitfor radiating low frequency spectral components and “satellite”loudspeaker units for radiating spectral components above thefrequencies radiated by the low frequency units. Low frequency units arereferred to by a number of names, including “subwoofers” “bass bins” andothers.

[0030] In surround sound systems having both an LFE channel and a basschannel, the LFE and bass channels may be combined and radiated by thelow frequency unit, as shown in FIG. 2a. In surround systems not havinga combined bass channel, each directional channel, including the bassportion of each directional channel) may be radiated by separatedirectional loudspeaker units, with only the LFE radiated by the lowfrequency unit. Still other surround systems may have more than one lowfrequency unit, one for radiating bass frequencies and one for radiatingthe LFE channel. “Full range” as used herein, refers to audible spectralcomponents having frequencies above those radiated by a low frequencyunit. If an audio system has no low frequency unit, “full range” refersto the entire audible frequency spectrum. “Directional channel” as usedherein is an audio channel that contains audio signals that are intendedto be transduced to sound waves that appear to come from a specificdirection. LFE channels and channels that have combined bass signalsfrom two or more directional channels are not, for the purposes of thisspecification, considered directional channels.

[0031] The directional channels, LS, L, LC, RC, R, and RS are processedby directional processor 36 to produce output audio signals at outputsignal lines 38 a-38 f for the acoustical drivers of the audio system.The signals output by directional processor 36 and the low frequencyunit signal in signal line 40 may then be further processed by systemequalization (EQ) and dynamic range control circuitry 42. (System EQ anddynamic range control circuitry is shown to illustrate the placement ofelements typical to audio processing circuitry, but does not perform afunction relevant to the invention. Therefore, system EQ and dynamicrange control circuitry 42 are not shown in subsequent figures and itsfunction will not be further described. Other audio processing elements,such as amplifiers that are not germane to the present invention are notshown or described). The directional channels are then transmitted tothe acoustical drivers for transduction to sound waves. The signal line38 a designated “left front (LF) array driver A” is directed toacoustical driver 12 of array 10 (of FIGS. 1a-1 c); the signal line 38 bdesignated “left front (LF) array driver B” is directed to acousticaldriver 11 of array 10 (of FIGS. 1a-1 c); the signal line38 c designated“right front (RF) array driver A” is directed to acoustical driver 17 ofarray 15 (of FIGS. 1a-1 c); and the signal line 38 d designated “rightfront (RF) array driver B” is directed to acoustical driver 16 of array15 (of FIGS. 1a-1 c). The signal line 38 e designated “left surround(LS) driver” is directed to limited range acoustical driver 22 of FIG.1b or acoustical driver 28 of FIG. 1c as will be explained below, andthe signal line 38 f designated “right surround (RS) driver” is directedto acoustical driver 24 of FIG. 1b or acoustical driver 30 of FIG. 1c,as will also be explained below. In some implementations, there is nooutput signal from LS output terminal 38 e or RS output terminal 38 f orboth. In other implementations one or both of LS output terminal 38 e orRS output terminal 38 f may be absent entirely, as will be explainedbelow.

[0032] Referring now to FIGS. 3a-3 d, there are shown four blockdiagrams of audio directional processor 36 for use with surround soundloudspeaker systems as shown in FIGS. 1a-1 c. FIGS. 3a-3 d show theportion of the directional processor for the LC, LS, and L channels. Ineach of the implementations, there is a mirror image for processing theRC, RS, and R channels. In FIGS. 3a-3 d, like reference numerals referto like elements performing like functions.

[0033]FIG. 3a shows the logical arrangement of directional processor 36for a configuration having no rear speakers. In FIG. 3a, the L channelis coupled to presentation mode processor 102 and to level detector 44.One output terminal 35 of presentation mode processor 102, designatedL′, is coupled to summer 47. The operation of presentation modeprocessor 102 will be described below in the discussion of FIG. 11. LSchannel is coupled to level detector 44 and frequency splitter 46. Leveldetector 44 provides front/rear scaler 48, front head related transferfunction (HRTF) filters and rear HRTF filters with signal levels tofacilitate the calculation of filter coefficients as will be describedbelow. Frequency splitter 46 separates the signal into a first frequencyband including signals below a threshold frequency and a secondfrequency band including signals above the threshold frequency. Thethreshold frequency is a frequency that corresponds to a wavelength thatapproximates dimensions of a human head. A convenient frequency is 2kHz, which corresponds to a wavelength of about 6.8 inches. Hereinafter,the portion of the surround signal above the threshold frequency will bereferred to as “high frequency surround signal” and the portion of thesurround signal below the threshold frequency will be referred to as“low frequency surround signal.” The low frequency surround signal isinput by signal path 43 to summer 54, or alternatively to summer 47 aswill be explained in the discussion of FIG. 3d. The high frequencysurround signal is input by signal path 45 to front/rear scaler 48,which splits the high frequency surround signal into a “front” portionand a “rear” portion in a manner that will be described below in thediscussion of FIG. 4. The “front” portion of the high frequency surroundsignal is transmitted by signal line 49 to front head related transferfunction (HRTF) filter 50, where it is modified in a manner that will bedescribed below in the discussion of FIG. 4. Modified front highfrequency surround is then optionally delayed by five ms by delay 52 andinput to summer 54. “Rear” portion of the high frequency surround signalis transmitted by signal line 51 to rear HRTF filter 56, where it ismodified in a manner that will be described below in the discussion ofFIG. 4. The modified rear portion is then optionally delayed by ten msby delay 58, and summed with front portion and low frequency surroundsignal at summer 54. The summed front, rear, and low frequency surroundportions are modified by front speaker placement compensator 60 (whichwill be further explained below following the discussion of FIGS. 4 and5) and input to summer 47, so that at summer 47 the L channel, the lowfrequency surround, and the modified high frequency surround are summed.The output signal of summer 47 may then be adjusted by a left/rightbalance control represented by multiplier 57 and is then inputsubtractively through time delay 61 to summer 62 and additively tosummer 58. LC channel is coupled to presentation mode processor 102.Output terminal 37, designated LC′ of presentation mode processor 102 iscoupled additively to summer 62 and subtractively through time delay 64to summer 58. Output signal of summer 58 is transmitted to acousticaldriver 11 (of FIGS. 1 and 2). Output signal of summer 62 is transmittedto acoustical driver 12 (of FIGS. 1 and 2). Time delays 61 and 64facilitate the directional radiation of the signals combined at summer47. If desired, the outputs of time delay 61 and 64 can be scaled by afactor such as 0.631 to improve directional radiation performance.Directional radiation using time delays is discussed in U.S. Pat. Nos.5,809,153 and 5,870,484 and will be further discussed below.

[0034]FIG. 3b shows directional processor 36 for a configuration havinga limited range rear speaker, that is, a speaker that is designed toradiate frequencies above the threshold frequency. In the circuitry ofFIG. 3b, summer 54 of FIG. 3a is not present. Instead, front HRTFfilters and optional five ms delay are coupled through front speakerplacement compensator 60 to summer 47 and rear HRTF filters. andoptional ten ms delay are coupled to rear speaker placement compensator66, which is in turn coupled to limited range acoustical driver 22 ofFIGS. 1 and 2.

[0035]FIG. 3c shows directional processor 36 for a configuration havinga full range rear speaker, that is, a speaker that is designed toradiate the full audible spectrum of frequencies above the frequenciesradiated by a low frequency unit. The circuitry of FIG. 3c is similar tothe circuitry of FIG. 3b, but low frequency surround signal output offrequency splitter 46 is summed with output signal of rear HRTF filterand optional ten ms delay 58 at summer 70, which is output to full-rangeacoustical driver 28.

[0036]FIG. 3d shows directional processor 36 that can be used with norear speaker, with a limited-range rear speaker, or with a full rangerear speaker. FIG. 3d includes a switch 68 and summer 69 arranged sothat with switch 68 in a closed position, the low frequency surroundsignal is directed to summer 70. With switch 68 in an open position, thelow frequency is directed to summer 47 for radiation from the frontspeaker array. FIG. 3d further includes a switch 72 and summer 73,arranged so that with switch 72 in an open position, the output signalfrom summer 70 is directed to rear speaker placement compensator 66 forradiation from a rear speaker. With switch 72 in a closed position, theoutput signal from summer 70 is directed to summer 54. With switch 72 inan open position and 68 in an open position, the circuitry of FIG. 3dbecomes the circuitry of FIG. 3b. With switch 72 in an open position andswitch 68 in a closed position, the circuitry of FIG. 3d becomes thecircuitry of FIG. 3c. With switch 72 in a closed position and switch 68in a closed position, the circuitry of FIG. 3d (since the effect of thesignal on line 43 being coupled to summer 54 as in the embodiment ofFIG. 3d is functionally equivalent to the signal on line 43 beingdirectly connected to summer 54 as in the embodiment of FIG. 3a) becomesthe circuitry of FIG. 3a. With switch 72 in a closed position and switch68 in an open position, the circuitry of FIG. 3d becomes the circuitryof FIG. 3a, with the low frequency surround signal directed to summer47.

[0037] In operation, switch 72 is set to the open position when there isa rear speaker and to the closed position when there is no rear speaker.Switch 68 is set to the open position for a limited range rear speakerand to the closed position for a full range rear speaker. Logically ifswitch 72 is set to the closed position, the position of switch 68should be irrelevant. It was stated in the preceding paragraph that thatif switch 72 is in the closed position, the low frequency surroundsignal may be summed with the high frequency surround signal before orafter the front speaker placement compensator depending on the positionof switch 68. However, as will be explained below in the discussion ofFIG. 4, the front and rear speaker placement compensators have littleeffect on frequencies below the threshold frequency, so it does notmatter whether the low frequency surround is summed with the highfrequency surround before or after the front speaker placementcompensator. Alternatively, switches 68 and 72 could be linked so thatif switch 72 is in the closed position, switch 68 would automatically beset to the open or closed position as desired.

[0038] In an exemplary embodiment, the directional processor 36 isimplemented as digital signal processors (DSPs) executing instructionswith digital-to-analog and analog-to-digital converters as necessary. Inother embodiments, the directional processor 36 may be implemented as acombination of DSPs, analog circuit elements, and digital-to-analog andanalog-to-digital converters as necessary.

[0039]FIG. 4, shows the frequency splitter 46, the front/rear scaler 48,the front HRTF filter 50 and the rear HRTF filter 56 of FIGS. 3a-3 c ingreater detail. Frequency splitter 46 is implemented as a high passfilter 74 and a summer 76. High pass filter 74 and summer 76 arearranged so that high pass filtered LS channel is combined subtractivelywith the LS channel signal so that the low frequency surround is outputon line 43. The high pass filter 74 is directly coupled to signal line45, so that the high frequency surround is output on signal line 45.Front/rear scaler is implemented as a summer 78 and a multiplier 80.Multiplier 80 scales the signal by a factor that is related to therelative amplitudes of the signals in the LS channel and the L channel.In the embodiment of FIG. 4, the factor is$\frac{\overset{\_}{LS}}{{\overset{\_}{LS}} + {\overset{\_}{L}}}.$

[0040] Summer 78 and multiplier 80 are arranged so that scaled signal iscombined subtractively with the unscaled signal and output on signalline 49 so that the signal on signal line 49 is the input signal scaledby$( {1 - \frac{\overset{\_}{LS}}{{\overset{\_}{LS}} + {\overset{\_}{L}}}} ).$

[0041] Multiplier is directly coupled to signal line 51 so that thesignal on the signal line 51 is the input signal scaled by$\frac{\overset{\_}{LS}}{{\overset{\_}{LS}} + {\overset{\_}{L}}}.$

[0042] It can be seen that if |{overscore (LS)}| approaches zero, theportion of the input signal that is directed to signal line 49approaches one and the portion of the signal that is directed to signalline 51 approaches zero. Similarly if |{overscore (LS)}| is much greaterthan |{overscore (L)}|, the portion of the input signal that is directedto signal line 49 approaches zero and the portion of the input signalthat is directed to signal line 51 approaches one. If |{overscore (LS)}|and |{overscore (L)}| are approximately equal, then the portion of theinput signal that is directed to signal line 49 is approximately equalto the portion of the input signal that is directed to signal line 51.The effect of the front/rear scaler is to orient the apparent source ofa sound relative to the listener. If |{overscore (L)}| is greater than|{overscore (LS)}|, a greater portion of the high frequency surroundsignal will be directed to the front speaker unit, and the apparentsource of the sound is toward the front. If |{overscore (LS)}| isgreater than |{overscore (L)}|, a greater portion of the high frequencysurround signal will be directed to the rear speaker unit (or in theabsence of a rear speaker unit, be processed so that it will appear tocome from the rear) and the apparent source of the sound is toward therear. If |{overscore (LS)}| and |{overscore (L)}| are relatively equal,then an approximately equal portion of the high frequency surroundsignal will be directed to the front and rear loudspeaker units, and theapparent source of the sound is to the side. The values |{overscore(L)}| and |{overscore (LS)}| are made available to multiplier 80 bylevel detectors 44 of FIGS. 3a-3 d. Scaling factors$\frac{\overset{\_}{LS}}{{\overset{\_}{LS}} + {\overset{\_}{L}}}$

[0043] and$( {1 - \frac{\overset{\_}{LS}}{{\overset{\_}{LS}} + {\overset{\_}{L}}}} )$

[0044] may be calculated as often as practical. In one implementation,the scaling factors are recalculated at five millisecond intervals.

[0045] Front HRTF filter 50 may be implemented as, in order in series, amultiplier 82, a first filter 84 representing the frequency shadingeffect of the head (hereinafter the head shading filter), a secondfilter 86 representing the diffraction path delay of the head(hereinafter the head diffraction path delay filter), a third filter 88representing the diffraction path delay of the pinna (hereinafter thepinna diffraction path delay filter), and a summer 90. Summer 90 sumsthe output signal from pinna diffraction path delay filter 88 with theoutput of head diffraction path delay filter 86, the output of headfrequency shading filter 84, and the unmultiplied input signal of frontHRTF filter 50. Rear HRTF filter 56 may be implemented as, in order inseries, multiplier 82, head frequency shading filter 84, pinnadiffraction path delay filter 88, head diffraction path delay 86, and afourth filter 92 representing the frequency shading effect of the rearsurface of the pinna (hereinafter the pinna rear frequency shadingfilter), and a summer 94. Summer 94 sums the output of pinna rearfrequency shading filter 92, output of head diffraction path delayfilter 86, pinna diffraction path delay filter 88, and the unmultipliedinput signal of the rear HRTF filter 56. In one implementation, thesignal from head diffraction path delay 86 to summer 94 is scaled by afactor of 0.5 and the signal from pinna rear frequency shading filter 92to summer 94 is scaled by a factor of two.

[0046] Head frequency shading filter 84 is implemented as a first orderhigh pass filter with a single real pole at −2.7 kHz; head diffractionpath delay filter 86 is implemented as a fourth order all-pass networkwith four real poles at −3.27 kHz and four real zeros at 3.27 kHz; pinnadiffraction delay filter 88 is implemented as a fourth order all-passnetwork with four real poles at −7.7 kHz and four real zeros at 7.7 kHz;and pinna rear frequency shading filter 92 is implemented as a firstorder high pass filter with a single real pole at −7.7 kHz. Multiplier82 scales the input signal by a factor of$\frac{Y}{( {Y - {\overset{\_}{LS}}} ) + ( {Y - {\overset{\_}{L}}} ) + Y},$

[0047] where Y is the larger of |{overscore (L)}| and |{overscore(LS)}|. The values |{overscore (L)}| and |{overscore (LS)}| are madeavailable to multiplier 80 by level detectors 44 of FIGS. 3a-3 d.“Pinna” as used herein refers to the auricle portion of the external earas shown on p. 1367 Gray's Anatomy, 38^(th) Edition, ChurchillLivingston 1995. “Pinna rear” or “rear surface of the pinna” as usedherein, refers to the anterior surface or the external ear, or theexternal ear as viewed in the direction of the arrow in Appendix 1. Thepinna is an acoustic surface for sounds from all directions, while therear pinna is an acoustic surface only for sounds from directionsranging from the side to the rear.

[0048] Filters having characteristics other than those described above(including a filter having a flat frequency response, such as a directelectrical connection) may be used in place of the filter arrangementsshown in FIG. 4 and described in the accompanying portion of thedisclosure.

[0049]FIG. 5 illustrates the purpose of the front speaker placementcompensator 60 and the rear speaker placement compensator 66 of FIGS.3a-3 d. Front speaker placement compensator is implemented as a filteror series of filters that has an effect that is inverse to the frontHRTF filter 50 when front HRTF filter 50 acts upon a signal thatradiated from a first specific angle. Similarly, the rear speakerplacement compensator is implemented as a filter or series of filtersthat has an effect that is inverse to the rear HRTF filter 56 when rearHRTF filter 56 acts upon a signal that radiated from a second specificangle.

[0050]FIG. 5 shows for explanation purposes a sound system according tothe configuration of FIG. 3b, with desired apparent source of a sound isat point Z, which is oriented at an angle θ relative to a listener 14.All angles in FIG. 5 lie in a horizontal plane which includes theentrances to the ear canals of listener 14. The reference line for theangles is a line passing through the points that are equidistant fromthe entrances to the ear canals of listener 14. Angles are measuredcounter-clockwise from the front of the listener 14. Placement of theapparent source of the sound at point Z is accomplished in part by thefront/rear scaler 48 of FIGS. 3a-3 c and FIG. 4. Front/rear scalerdirects more of the high frequency surround signal to the front array 10than to the rear speaker unit, so that the apparent source of the soundis somewhat forward. Placement of the apparent source of the sound atpoint Z is further accomplished by the front and rear HRTF filters 50and 56 (of FIGS. 3a-3 d) respectively. Front and rear HRTF filters 50and 56 alter the audio signals so that when the signals are transducedto sound waves by front array 10 and limited range acoustical driver 22,the sound waves will have the frequency content and phase relationshipsas if the sound waves had originated at point Z and had been modified bythe head 96 and pinna 98 of listener 14. However, when the sound wavesare actually transduced by front array 10 and rear limited rangeacoustical driver 22, the frequency content and the phase relationshipsof the sound waves will be modified by the physical head 96 and pinna 98of listener 14, so that in effect the sound waves that reach the earcanal have the frequency content and phase relationships that have beentwice modified by the head and pinna of the listener over angle φ₁.Front speaker placement compensator 60 modifies the audio signal so thatwhen it is transduced by front array 10, the sound waves will not havethe change in frequency content and phase relationships attributable tothe angle φ₁, leaving in the audio signal the change in frequency andphase relationships attributable to the difference between angle θ andangle φ₁. Then, when the sound waves are transduced by front array 10and modified by the head and pinna of the listener, the sound waves thatreach the ear canal will have the frequency content and phaserelationships as a sound from a source at angle θ. Similarly, the rearspeaker placement compensator 66 modifies the audio signal so that whenit is transduced by rear limited range acoustical driver 22, the soundwaves will not have the change in frequency content and phaserelationships attributable to the angle φ₂, leaving the change infrequency and phase relationships attributable to the difference betweenangle θ and angle φ₂. Then, when the sound is transduced by rear limitedrange acoustical driver 22, the sound waves that reach the ear canalwill have the same frequency content and phase relationships as a soundfrom a source at angle θ. If the speaker configuration is theconfiguration of FIG. 3a the same explanation applies. However theconfiguration having the limited range rear speaker was chosen toillustrate that the front and rear HRTF filters 50 and 56 and the frontand rear speaker placement compensators 60 and 66, all have littleeffect below frequencies having corresponding wavelengths thatapproximate the dimensions of the head, for example 2 kHz. In oneembodiment, the angles φ₁ and φ₂ are measured and input into audiosystem so that speaker placement compensators 60 and 66 calculate usingthe precise angle. One technique for measuring angles φ₁ and φ₂ is tophysically measure them. In a second embodiment, speaker placementcompensators are set to pre-selected typical values of angles φ₁ and φ₂(for example 30 degrees and 150 degrees). This second embodiment givesacceptable results, but does not require actual measurement of thespeaker placement angles and may require somewhat less complex computingin speaker placement compensators 60 and 66.

[0051] Speaker placement compensators 60 and 66 may be implemented asfilters having the inverse effect as front and rear HRTF filters,respectively, evaluated for the selected values of angles φ₁ and φ₂, byusing values derived from the relationships${\varphi_{1} = {{{\arcsin \lbrack {1 - \lbrack \frac{Y - {\overset{\_}{LS}} + Y - {\overset{\_}{L}}}{Y} \rbrack} \rbrack}\quad {and}\quad \varphi_{2}} = {\arcsin \lbrack {1 - \lbrack \frac{Y - {\overset{\_}{LS}} + Y - {\overset{\_}{L}}}{Y} \rbrack} \rbrack}}},$

[0052] respectively.

[0053] If some filter arrangement other than the filter arrangement ofFIG. 4 is used for the front HRTF filter 50 and the rear HRTF filter 56,the front speaker placement compensator 60 and the rear speakerplacement compensator 66 may be modified accordingly. If HRTF filters 50and 56 have a flat frequency response, the front speaker placementcompensator 60 and rear speaker placement compensator 66 may be replacedby a filter having a flat frequency response (such as a directelectrical connection).

[0054] Referring now to FIG. 6, there is shown an example of two moreacoustical loudspeaker configurations for illustrating another featureof the invention. In FIG. 6, there is an acoustical driver array 10,similar to the acoustical driver array 10 of FIGS. 1a-1 c, placed at apoint displaced by 30 degrees from listener 14. In addition, there arelimited range acoustical drivers, similar to the limited rangeacoustical drivers 22 of FIGS. 1a-1 c, at 60 degrees, 90 degrees, 120degrees, and 150 degrees OR full range acoustical drivers 28 similar tothe full range acoustical drivers 28 of FIGS. 1a-1 c. The limited rangeacoustical drivers are designated 22-60, 22-90, 22-120, and 22-150,respectively, to indicate the angular position of the limited rangeacoustical driver. The alternate full range acoustical drivers aredesignated 28-60, 28-90, 28-120, and 28-150, respectively, to indicatethe angular position of the limited range acoustical driver. All anglesin FIG. 6 lie in the horizontal plane that includes the entrances to theear canal of listener 14. The reference line for the angles is a linepassing through the points that are equidistant from the entrances tothe listener's ear canals. The angles for the acoustical driver units onthe left of listener 14 are measured counterclockwise from the referenceline in front of the listener. The angles for the acoustical driverunits on the right of listener 14 are measured clockwise from thereference line in front of the listener. There may also be otheracoustical driver units, such as a center channel acoustical driver unitor a low frequency unit, which are not shown in this view.

[0055]FIG. 7 shows a block diagram of an audio signal processing systemfor providing audio signals for the loudspeaker units of FIG. 6. Anaudio signal source 32 is coupled to a decoder 34 which decodes theaudio source from the audio signal source into a plurality of channels,in this case a low frequency effects (LFE) channel, and bass channel,and a number of directional channels, including a left (L) channel, aleft center (LC) channel, and further including a number of leftchannels, L60, L90, L120, and LS in which the numerical indicatorcorresponds to the angular displacement, in degrees, of the channelrelative to the listener. There are corresponding right channels, RC, R,R60, R90, R120 and RS. The remainder of the discussion will focus on theleft channels, since the right channels can be processed in a similarmanner to the left channels. The left channel signals are processed bydirectional processor 36 to produce output signals for low frequency(LF) array driver 12 on signal line 38 a, for LF array driver 11 onsignal line 38 b, for driver 22-60L or driver 28-60L on signal line 39a, for driver 22-90L or driver 28-90L on signal line 39 b, for driver22-120L or 28-120L on signal line 39 c, and for driver 22-150L or driver28-150L on signal line 39 d. As with the embodiment of FIG. 2a, theoutputs on the signal lines are processed by system EQ and dynamic rangecontroller 42.

[0056] In an exemplary embodiment, the directional processor 36 isimplemented as digital signal processors (DSPs) executing instructionswith digital to analog and analog-to-digital converters as necessary. Inother embodiments, the directional processor 36 may be implemented as acombination of DSPs, analog circuit elements, and digital to analog andanalog-to-digital converters as necessary.

[0057]FIG. 8 shows a block diagram of the directional processor 36 ofFIG. 7, for an implementation with limited range side and rearacoustical drivers. The directional processor has inputs for five leftdirectional channels. The five directional channels can be created froman audio signal processing system having two channels, a left (L)channel designed, for example, to be radiated at 30 degrees) and a leftsurround (LS) channel, designed, for example to be radiated at 150degrees). The L and LS channels can be decoded according the teachingsof U.S. patent application Ser. No. 08/796,285, incorporated herein byreference, to produce channel L90 (intended to be radiated at 90degrees). Channels L and L90 and channels L90 and LS can then be decodedto produce channels L60 and L120, respectively. The invention will workequally well with fewer directional channels or more directionalchannels. The audio signal processing system of FIG. 7 has severalelements that are similar to elements of the system of FIGS. 3a-3 d andperform similar functions to the corresponding elements of FIGS. 3a-3 d.The similar elements use similar reference numerals. Some elements ofFIGS. 3a-3 d that are not germane to the invention (such as multiplier57) are not shown in FIG. 8. A mirror image audio processing systemcould be created to process right directional channels corresponding tothe left directional channels.

[0058] Referring now to FIG. 8, the input terminals for channels L60,L90, L120, and LS are coupled to level detector 44 for makingmeasurements for the scalers and HRTF filters. The input terminal forchannel L is coupled to presentation mode processor 102. Output terminal35 designated L′ of presentation mode processor 102 is coupled to summer47. The input terminal for channel LC is coupled to presentation modeprocessor 102. Output terminal 37 of presentation mode processor 102designated LC′ is coupled subtractively to summer 58 through time delay58 and additively to summer 62. The audio signal in channel L60 is splitby frequency splitter 46a into a low frequency (LF) portion and a highfrequency (HF) portion. LF portion is input to summer 47. HF portion ofthe audio signal in channel L60 is input to front/rear scaler 48 a,(similar to the front/rear scaler 48 of FIGS. 3a-3 d and 4), using thevalues |{overscore (L)}| and |{overscore (L60)}| respectively for thevalues |{overscore (L)}| and |{overscore (LS)}| in the discussion ofFIG. 4. Front/rear scaler 48 a separates the HF portion of the audiosignal in channel L60 into a “front” portion and a “rear” portion. Frontportion of the HF portion of the audio signal in channel L60 isprocessed by front HRTF filter 50 a (similar to the front HRTF filter 50of FIGS. 3a-3 d and 4), using the values |{overscore (L)}| and|{overscore (L60)}| respectively for the values |{overscore (L)}| and|{overscore (LS)}| in the discussion of FIG. 4, and speaker placementcompensator 60 a, (similar to the speaker placement compensator 60 ofFIGS. 3a-3 d and 4), calculated for 30 degrees, and input to summer 47.Rear portion of the audio signal in channel L60 is processed by frontHRTF filter 50 b (similar to the front HRTF filter 50 of FIGS. 3a-3 dand 4), using the values |{overscore (L)}| and |{overscore (L60)}|respectively for the values |{overscore (L)}| and |{overscore (LS)}| inthe discussion of FIG. 4) and speaker placement compensator 60 a,similar to the speaker placement compensator 60 of FIGS. 3a-3 d and 4,calculated for 60 degrees, and input to summer 100-60.

[0059] The audio signal in channel L90 is split by frequency splitter 46b into a low frequency (LF) portion and a high frequency (HF) portion.LF portion is input to summer 47. HF portion of the audio signal inchannel L90 is input to front/rear scaler 48 b, similar to thefront/rear scaler 48 of FIGS. 3a-3 d and 4, using the values |{overscore(L60)}| and |{overscore (L90)}| respectively for the values |{overscore(L)}| and |{overscore (LS)}| in the discussion of FIG. 4. Front/rearscaler 48 b separates the HF portion of the audio signal in channel L90into a “front” portion and a “rear” portion. Front portion of the HFportion of the audio signal in channel L90 is processed by front HRTFfilter 50 c (similar to the front HRTF filter of FIGS. 3a-3 d and 4),using the values |{overscore (L60)}| and |{overscore (L90)}|respectively for the values |{overscore (L)}| and |{overscore (LS)}| inthe discussion of FIG. 4), and speaker placement compensator 60 b,calculated for 60 degrees, and input to summer 100-60. Rear portion ofthe audio signal in channel L60 is processed by front HRTF filter 50 d(similar to the front HRTF filter of FIGS. 3a-3 d and 4), using thevalues |{overscore (L60)}| and |{overscore (L90)}| respectively for thevalues |{overscore (L)}| and |{overscore (LS)}| in the discussion ofFIG. 4, and speaker placement compensator 60 d, (similar to the speakerplacement compensator 60 of FIGS. 3a-3 d and 4), calculated for 90degrees, and input to summer 100-90.

[0060] The audio signal in channel L120 is split by frequency splitter46 c into a low frequency (LF) portion and a high frequency (HF)portion. LF portion is input to summer 47. HF portion of the audiosignal in channel L120 is input to front/rear scaler 48c, (similar tothe front/rear scaler 48 of FIGS. 3a-3 d and 4), using the values|{overscore (L90)}| and |{overscore (L120)}| respectively for the values|{overscore (L)}| and |{overscore (LS)}| in the discussion of FIG. 4.Front/rear scaler 48c separates the HF portion of the audio signal inchannel L120 into a “front” portion and a “rear” portion. Front portionof the HF portion of the audio signal in channel L120 is processed byfront HRTF filter 50 e (similar to the front HRTF filter 50 of FIGS.3a-3 d and 4, using the values |{overscore (L90)}| and |{overscore(L120)}| respectively for the values |{overscore (L)}| and |{overscore(LS)}| in the discussion of FIG. 4 and speaker placement compensator 60e (similar to the speaker placement compensator 60 of FIGS. 3a-3 d and4), calculated for 90 degrees, and input to summer 100-90. Rear portionof the audio signal in channel |{overscore (L90)}| is processed by rearHRTF filter 56 a (similar to the rear HRTF filter 56 of FIGS. 3a-3 d and4), using the values |{overscore (L90)}| and |{overscore (L120)}|respectively for the values |{overscore (L)}| and |{overscore (LS)}|,and speaker placement compensator 60 f (similar to the speaker placementcompensator 60 of FIGS. 3a-3 d and 4), calculated for 120 degrees, andinput to summer 100-120.

[0061] The audio signal in channel LS is split by frequency splitter 46d into a low frequency (LF) portion and a high frequency (HF) portion.LF portion is input to summer 47. HF portion of the audio signal inchannel LS is input to front/rear scaler 48 d, (similar to thefront/rear scaler 48 of FIGS. 3a-3 d and 4), using the values|{overscore (L120)}| and |{overscore (LS)}| respectively for the values|{overscore (L)}| and |{overscore (LS)}| in the discussion of FIG. 4.Front/rear scaler 48 d separates the HF portion of the audio signal inchannel LS into a “front” portion and a “rear” portion. Front portion ofthe HF portion of the audio signal in channel LS is processed by rearHRTF filter 56 b (similar to the rear HRTF filter 56 of FIGS. 3a-3 d and4), using the values |{overscore (L120)}| and |{overscore (LS)}|respectively for the values |{overscore (L)}| and |{overscore (LS)}| inthe discussion of FIG. 4, and speaker placement compensator 60 fg(similar to the speaker placement compensator 60 of FIGS. 3a-3 d and 4),calculated for 120 degrees, and input to summer 100-120. Rear portion ofthe audio signal in channel LS is processed by rear HRTF filter 56 c(similar to the rear HRTF filter 56 of FIGS. 3a-3 d and 4), and speakerplacement compensator 60 h (similar to the speaker placement compensator60 of FIGS. 3a-3 d and 4), calculated for 150 degrees.

[0062] The output signal of summer 47 is transmitted additively tosummer 58 and subtractively through time delay 61 to summer 62. Theoutput signal of summer 58 is transmitted to full range acousticaldriver 11 (of speaker array 10) for transduction to sound waves. Theoutput signal of summer 62 is transmitted to full range acousticaldriver 12 for transduction to sound waves. Time delay 61 facilitates thedirectional radiation of the signals combined at summer 47. Outputsignals of summers 100-60, 100-90, 100-120, and of speaker placementcompensator 60 h are transmitted to limited range acoustical drivers22-60, 22-90, 22-120, and 22-150, respectively, for transduction tosound waves.

[0063]FIG. 9 shows the directional processor of FIG. 7 for animplementation having full range side and rear acoustical drivers. Theimplementation of FIG. 9 has the same input channels as theimplementation of FIG. 7. The invention will work with fewer directionalchannels or more directional channels. The audio signal processingsystem of FIG. 7 has several elements that are similar to elements ofthe system of FIGS. 3a-3 d and perform similar functions to thecorresponding elements of FIGS. 3a-3 d. The similar elements use similarreference numerals. A mirror image audio processing system could becreated to process right directional channels corresponding to the leftdirectional channels.

[0064]FIG. 9 is similar to FIG. 8, except for the following. The lowfrequency (LF) signal line from frequency splitter 46a is coupled tosummer 100-60 instead of summer 47; the LF signal line from frequencysplitter 46 b is coupled to summer 100-90 instead of summer 47; the LFsignal line from frequency splitter 46 c is coupled to summer 100-120instead of summer 47; the LF signal line from frequency splitter 46 d iscoupled to summer 100-150 instead of summer 47; and the output ofspeaker placement compensator 60 h is coupled to a summer 100-150.Output signals of summers 100-60, 100-90, 100-120, and 100-150 aretransmitted to full range acoustical drivers 28-60, 28-90, 28-120, and28-150, respectively, for transduction to sound waves.

[0065] Referring now to FIGS. 10a-10 c, there are shown three topdiagrammatic views of some of the components of an audio system fordescribing another feature of the invention. As described in patentssuch as U.S. Pat. Nos. 5,809,153 and 5,870,484, arrays of acousticaldrivers and signal processing techniques can be designed to radiatesound waves directionally. By radiating the same sound wave from twoacoustical drivers subtractively (functionally equivalent to out ofphase) and time-delayed, a radiation pattern can be created in which theacoustic output is greatest along one axis (hereinafter the primaryaxis) and in which the acoustic output is minimized in another direction(hereinafter the null axis). In FIGS. 10a-10 c, an array 10, includingacoustical drivers 11 and 12 is arranged as in an audio system shown inFIGS. 1a-1 c, 2 a, and FIGS. 3a-3 d. The parameters of time delay 64 ofFIGS. 3a-3 d are set such that a signal that is transmitted undelayed toacoustical driver 12 and delayed to acoustical driver 11 and transducedresults in a radiation pattern that has a primary axis in a direction104 generally toward a listener 14 in a typical listening position, anull axis in a direction 106 generally away from listener 14 in atypical listening position, and a radiation pattern 105 as indicated insolid line. The parameters of time delay 61 of FIGS. 3a-3 d are set suchthat a signal that is transmitted undelayed to acoustical driver 11 anddelayed to acoustical driver 12 and transduced results in a radiationpattern that has a primary axis in direction 106 generally away from alistener 14 in a typical listening position, a null axis in direction104 generally toward listener 14 in a typical listening position, and aradiation pattern 107 as indicated in dashed line. In FIG. 10a, theaudio signal in channel LC is processed and radiated such that theradiation pattern has a primary axis in direction 104 and a null axis indirection 106 and the audio signal in channels L and LS are processedand radiated such that they have a primary axis in direction 106. InFIG. 1b, the audio signal in channels L and LC are processed andradiated such that the radiation patterns have a primary axis indirection 104 and a null axis in direction 106, and the audio signal inchannel LS is processed and radiated such that it has a primary axis indirection 106 and a null axis in direction 104. In FIG. 10c, the audiosignals in channels L, LC, and LS are processed and radiated such thatthey all have primary axes in direction 106 and null axes in direction104. Hereinafter, the combination of radiation patterns, primary axes,and null axes will referred to as “presentation modes.” Generally, thepresentation mode of FIG. 10a is preferable when the audio system isused as a part of a home theater system, in which is desirable to have astrong center acoustic image and a “spacious” feel to the directionalchannels. The presentation mode of FIG. 10b may be preferable when theaudio system is used to play music, when center image is not soimportant. The presentation mode of FIG. 10c may be preferable if theaudio system is placed in a situation in which the array 10 must beplaced very close to a center line (that is when the angle φ₁ of FIG. 5is small). As with several of the previous figures, there may be mirrorimage audio system for processing the right side directional channels.

[0066] Referring now to FIG. 11, there is shown presentation modeprocessor 102 (of FIGS. 3a-3 c, 8, and 9) in more detail. Channel Linput is connected additively to summer 108 and to the one side ofswitch 110. Other side of switch 110 is connected additively to summer112 and subtractively to summer 108. Channel LC is connected additivelyto summer 112 which is connected additively to summer 116 and to oneside of switch 118. Other side of switch 118 is connected additively tosummer 114 and subtractively to summer 116. Summer 114 is connected toterminal 35, designated L′. Summer 116 is connected to terminal 37,designated LC′. Depending on whether switches 110 and 118 are in theopen or closed position, the signal at output terminal 35 (designatedL′) may be the signal that was input from channel L, the combined inputsignals from channels L and LC, or no signal. Depending on whetherswitches 110 and 118 are in the open or closed position, the signal atoutput terminal 37 (designated LC′) may be the signal that was inputfrom channel LC, the combined input signals from channels L and LC, orno signal.

[0067] Referring now to any of FIGS. 3a-3 c, the output signal ofterminal 35 is summed with the low frequency portion of the surroundchannel at summer 47, and is transmitted to summer 58, which is coupledto acoustical driver 11, and through time delay 61 to summer 62, whichis coupled to acoustical driver 12. The output signal of terminal 37 iscoupled to summer 62 and through time delay 64 to summer 58. Thus theoutput of terminal 35 is summed with the low frequency (LF) portion ofthe left surround (LS) signal and transmitted undelayed to acousticaldriver 11 and delayed to acoustical driver 12. The output of terminal 37is transmitted undelayed to acoustical driver 12 and delayed toacoustical driver 11. As taught above in the discussion of FIGS. 10a-10c, the parameters of time delay 64 may be set so that an audio signalthat is transmitted undelayed to acoustical driver 12 and delayed toacoustical driver 11 and transduced results in an radiation pattern thathas a primary axis in direction 104 of FIGS. 10a-10 b. Similarly, thediscussion of FIGS. 10a-10 c teaches that the parameters of time delay61 may be set so that an audio signal that is transmitted undelayed toacoustical driver 11 and delayed to acoustical driver 12 and transducedresults in radiation pattern that has a primary axis in direction 106 ofFIGS. 10a-10 b. Therefore, by setting the switches 110 and 118 ofpresentation mode processor 102 to the “closed” or “open” position, itis possible for a user to achieve the presentation modes of FIGS. 10a-10c. The table below the circuit of FIG. 11 shows the effect of thevarious combinations of “open” and “closed” positions of switches 110and 118. For each of the four combinations, the table shows which ofchannels L and LC are output on the output terminals designated L′ andLC′ (terminals 35 and 37, respectively), which channels when radiatedhave a radiation pattern that has a primary axis in direction 104 and anull axis in direction 106 and which have a primary axis in direction106 and a null axis in direction 104, and which of FIGS. 10a-10 c areachieved by the combination of switch settings. In the implementation ofFIGS. 3a-3 c, 10, and 11, the low frequency portion of surround channelLS is always radiated with the primary axis in direction 106. Also, ifswitch 118 is in the closed position, the radiation pattern of FIG. 10cresults, regardless of the position of switch 110.

[0068] In the implementations of FIGS. 8 and 9, the presentation modeprocessor 102 has the same effect on input channels L and LC and thesignals on the output terminals 35 and 37 (designated L′ and LC′,respectively).

[0069] It is evident that those skilled in the art may now make numerousmodifications of and departures from the specific apparatus andtechniques herein disclosed without departing from the inventiveconcepts. Consequently, the invention is to be construed as embracingeach and every novel feature and novel combination of features hereindisclosed and limited only by the spirit and scope of the appendedclaims.

What is claimed is:
 1. In an audio system having a first audio signaland a second audio signal, said first and second audio signals havingamplitudes, a method for processing said audio signals, comprising:dividing said first audio signal into a first spectral band signal and asecond spectral band signal; scaling said first spectral band signal bya first scaling factor to create a first signal portion, wherein saidfirst scaling factor is proportional to said amplitude of said secondaudio signal; and scaling said first spectral band signal by a secondscaling factor to create a second signal portion.
 2. A method forprocessing audio signals in accordance with claim 1, wherein said secondscaling factor is proportional to said amplitude of said first audiosignal.
 3. A method for processing audio signals in accordance withclaim 1, wherein said first and second audio signals are associated withdirectional channels in a multichannel audio system.
 4. A method forprocessing audio signals in accordance with claim 3, further comprising,filtering said first signal portion by a first filter to produce afiltered first signal portion, and filtering said second signal portionby a second filter to produce a filtered second signal portion.
 5. Amethod for processing audio signals in accordance with claim 4, wherein${\frac{SF1}{SF2} = \frac{ampl2}{ampl1}},$

wherein SF1 is said first scaling factor, SF2 is said second scalingfactor, ampl1 is said amplitude of said first audio signal and ampl2 issaid amplitude of said second audio signal.
 6. A method for processingaudio signals in accordance with claim 5, wherein said first filter andsaid second filter include a filter portion having a frequency responseand time delay effect similar to that of the human head.
 7. A method forprocessing audio signals in accordance with claim 5, further comprisingcombining said filtered first signal portion with said second audiosignal.
 8. A method for processing audio signals in accordance withclaim 5, further comprising combining said filtered second signalportion with said second spectral band signal.
 9. A method forprocessing audio signals in accordance with claim 5, further comprisingcombining said filtered first signal portion, said filtered secondsignal portion and said second spectral band signal.
 10. A method forprocessing audio signals in accordance with claim 4, further comprisingthe step of combining said filtered first signal portion with saidsecond audio signal.
 11. A method for processing audio signals inaccordance with claim 4, further comprising combining said filteredsecond signal portion with said second spectral band signal.
 12. Amethod for processing audio signals in accordance with claim 4, furthercomprising the step of combining said filtered first signal portion,said filtered second signal portion and said second spectral bandsignal.
 13. A method for processing audio signals in accordance withclaim 1, wherein ${\frac{SF1}{SF2} = \frac{ampl2}{ampl1}},$

wherein SF1 is said first scaling factor, SF2 is said second scalingfactor, ampl1 is said amplitude of said first audio signal and ampl2 issaid amplitude of said second audio signal.
 14. A method for processingaudio signals in accordance with claim 1, further comprising, filteringsaid first signal portion by a first filter to produce a filtered firstsignal portion, and filtering said second signal portion by a secondfilter to produce a filtered second signal portion.
 15. A method forprocessing audio signals in accordance with claim 14, wherein said firstfilter and said second filter include a filter portion having afrequency response and time delay effect similar to that of the humanhead.
 16. A method for processing audio signals in accordance with claim15, wherein one of said first filter or said second filter has filterportion having a frequency response and time delay effect similar tofrequency response and time delay effect of the human head on a soundwave arriving from the front of said human head and the other of saidfirst filter or second filter has filter portion having a frequencyresponse and time delay effect similar to frequency response and timedelay effect of the human head on a sound wave arriving from the rear ofsaid human head.
 17. A method for processing audio signals in accordancewith claim 15, wherein said first filter and said second filter have afilter portion having frequency response and time delay effect similarto frequency response and time delay effect of the human head on a soundwave arriving from the front of said human head.
 18. A method forprocessing audio signals in accordance with claim 15, wherein said firstfilter and said second filter have a filter portion having a frequencyresponse and time delay effect similar to frequency response and timedelay effect of the human head on a sound wave arriving from the rear ofsaid human head.
 19. A method for processing audio signals in accordancewith claim 15, wherein said first filter and said second filter includea filter portion having a frequency response and time delay effectinverse to said filter having a frequency response and time delay effectsimilar to the human head.
 20. A method for processing audio signals inaccordance with claim 14, wherein one of said first filter or saidsecond filters has a flat frequency response.
 21. A method forprocessing audio signals in accordance with claim 20, wherein the otherof said first filter or said second filters has a flat frequencyresponse.
 22. A method for processing audio signals in accordance withclaim 14, further comprising, combining said filtered first signalportion with said second audio signal to produce a first combinedsignal.
 23. A method for processing audio signals in accordance withclaim 22, with an audio system including a directional loudspeaker unit,said combining further including combining said second spectral band andsaid filtered second signal portion so that said first combined signalincludes said filtered first signal portion, said filtered second signalportion, said second spectral band, and said second audio signal andfurther comprising, electroacoustically transducing, by said directionalloudspeaker unit, said first combined signal.
 24. A method forprocessing audio signals in accordance with claim 22, with an audiosystem further including a directional loudspeaker unit and aloudspeaker unit distinct from said directional loudspeaker unit andfurther comprising, combining said second spectral band and saidfiltered second signal portion to produce a second combined signal;electroacoustically transducing, by said loudspeaker unit, said secondcombined signal; and electroacoustically transducing, by saiddirectional loudspeaker unit, said first combined signal.
 25. A methodfor processing audio signals in accordance with claim 22 with an audiosystem including a directional loudspeaker unit and a loudspeaker unitdistinct from said directional loudspeaker unit, said distinctloudspeaker unit substantially limited to radiating spectral componentsin said first spectral band, said combining further comprising,combining said second spectral band signal so that said first combinedsignal includes said filtered first signal portion, said second spectralband signal, and said second audio signal, said method furthercomprising, electroacoustically transducing, by said directionalloudspeaker unit, said first combined signal; and electroacousticallytransducing, by said loudspeaker unit, said filtered second signalportion.
 26. A method for processing audio signals in accordance withclaim 1, wherein said first scaling factor and said second scalingfactor are variable with respect to time.
 27. A method for processingaudio signals in accordance with claim 1, wherein the sum of said firstscaling factor and said second scaling factor is one.
 28. In an audiosystem having a first audio signal, a second audio signal and adirectional loudspeaker unit, a method for processing said audio signalscomprising, electroacoustically directionally transducing said firstaudio signal to produce a first signal radiation pattern;electroacoustically directionally transducing said second audio signalto produce a second signal radiation pattern; wherein said first signalradiation pattern and said second signal radiation pattern arealternatively and user selectively similar or different.
 29. A methodfor processing audio signals in accordance with claim 28 with an audiosystem including a source of a third audio signal and a speaker unitseparate from said directional loudspeaker unit further comprising,electroacoustically transducing said third audio signal by said speakerunit.
 30. A method for processing audio signals in accordance with claim29, wherein said third audio signal is substantially limited to afrequency range having a lower limit at a frequency that has acorresponding wavelength that approximates the dimensions of a humanhead and wherein said speaker unit is constructed and arranged toelectroacoustically transduce audio signals having frequencies in saidfrequency range.
 31. A method for processing audio signals in accordancewith claim 30, wherein said third audio signal comprises a firstspectral band of a scaled, filtered audio signal representing adirectional channel of a multichannel audio system.
 32. A method forprocessing audio signals in accordance with claim 29, wherein said thirdaudio signal comprises a filtered scaled first spectral band of an inputaudio signal representing a directional channel of a multichannel audiosystem and a second spectral band of said input audio signal.
 33. In anaudio system having a first audio signal, a second audio signal, a thirdaudio signal that is substantially limited to a frequency range having alower limit at a frequency that has a corresponding wavelength thatapproximates the dimensions of a human head, a directional loudspeakerunit, and a loudspeaker unit, distinct from said directional loudspeakerunit, a method for processing said audio signals comprising,electroacoustically directionally transducing by said directionalloudspeaker unit said first audio signal to produced a first radiationpattern; electroacoustically directionally transducing by saiddirectional loudspeaker unit said second audio signal to produce asecond radiation pattern; and electroacoustically transducing by saiddistinct loudspeaker unit said third audio signal.
 34. A method forprocessing audio signals in accordance with claim 33, wherein saidelectroacoustically directionally transducing compriseselectroacoustically directionally transducing said first audio signal sothat said first radiation pattern has a primary axis in a firstdirection and so that said second radiation pattern has a primary axisin a second direction different from said first direction.
 35. A methodfor processing audio signals in accordance with claim 33, wherein saidthird audio signal comprises a first spectral band of a scaled, filteredaudio signal representing a directional channel of a multichannel audiosystem.
 36. In an audio system having a plurality of directionalchannels, a method for processing audio signals respectivelycorresponding to each of said plurality of channels, comprising,dividing a first audio signal into a first audio signal first spectralband signal and a first audio signal second spectral band signal;scaling said first audio signal first spectral band signal by a firstscaling factor to create a first audio signal first spectral band firstportion signal; scaling said first audio signal first spectral bandsignal by a second scaling factor to create a first audio signal firstspectral band second portion signal; dividing a second audio signal intoa second audio signal first spectral band signal and a second audiosignal second spectral band signal; scaling said second audio signalfirst spectral band signal by a third scaling factor to create a secondaudio signal first spectral band first portion signal; and scaling saidsecond audio signal first spectral band signal by a fourth scalingfactor to create a second audio signal first spectral band secondportion signal.
 37. A method for processing audio signals, in accordancewith claim 36, further comprising, filtering said first audio signalfirst spectral band first portion signal by a first filter to produce afiltered first audio signal first spectral band first portion signal,filtering said first audio signal first spectral band second portionsignal by a second filter to produce a filtered first audio signal firstspectral band second portion signal, filtering said second audio signalfirst spectral band first portion signal by a third filter to produce afiltered second audio signal first spectral band first portion signal,and filtering said second audio signal first spectral band first portionsignal by a fourth filter to produce a filtered second audio signalfirst spectral band first portion signal.
 38. A method for processingaudio signals in accordance with claim 37 with an audio system having adirectional loudspeaker unit, and a first loudspeaker unit and a secondloudspeaker unit, both distinct from said directional loudspeaker unitand distinct from each other, said first and second distinct loudspeakerunits substantially limited to radiating frequencies in said firstspectral band, wherein said spectral band has a lower frequency limitthat corresponds to a wavelength approximating the dimensions of thehuman head, said method further comprising, combining said first audiosignal second spectral band signal, said second audio signal secondspectral band, and a third audio signal to produce a first combinedsignal; electroacoustically transducing by said directional loudspeakerunit, said first combined signal; combining said filtered first audiosignal first spectral band second portion with said filtered secondaudio signal first spectral band second signal first portion to producea second combined signal; electroacoustically transducing by said firstdistinct loudspeaker unit said second combined signal; andelectroacoustically transducing by said second distinct loudspeakerunit, said filtered second audio signal first spectral band secondportion.
 39. A method for processing audio signals in accordance withclaim 38, further comprising, combining said filtered second audiosignal first spectral band second portion signal with a filtered,spectral band-limited portion of a signal representing an adjacentchannel to produce a third combined signal; and electroacousticallytransducing by said second distinct loudspeaker unit, said thirdcombined signal.
 40. A method for processing audio signals in accordancewith claim 37 with an audio system having a directional loudspeakerunit, a first loudspeaker unit distinct from said directionalloudspeaker unit, and a second loudspeaker unit distinct from saiddirectional loudspeaker unit and said first distinct loudspeaker unit,said method further comprising, combining a third of said plurality ofaudio signals and said filtered first audio signal first spectral bandfirst portion to produce a first combined audio signal;electroacoustically transducing by said directional loudspeaker unitsaid first combined signal; combining said filtered second audio signalfirst spectral band first portion, said filtered first audio signalfirst spectral band second portion, and said first audio signal secondspectral band to produce a second combined signal; electroacousticallytransducing by said first distinct loudspeaker unit said second combinedsignal; combining said filtered second audio signal first spectral bandsecond portion and said second audio signal second spectral band signalto produce a third combined signal; and electroacoustically transducingby said second distinct loudspeaker unit said third combined signal. 41.A method for processing audio signals in accordance with claim 40,further comprising, combining said filtered second audio signal firstspectral band second portion signal with a filtered, spectral bandlimited portion of a signal representing an adjacent channel to producea third combined signal; and electroacoustically transducing by saidsecond distinct loudspeaker unit, said third combined signal.
 42. Amethod for processing an audio signal, comprising, filtering said audiosignal by a first filter, said first filter having a frequency responseand time delay effect similar to the human head to produce aonce-filtered audio signal; filtering said once-filtered audio signal bya second filter, said second filter having a frequency response and timedelay effect inverse to the frequency and time delay effect of a humanhead on a sound wave.
 43. A method for processing audio signals inaccordance with claim 42, wherein said second filter has a time delayeffect inverse to the frequency and time delay effect of a human head ona sound wave that originates at a preselected orientation relative tosaid human head.
 44. A method for processing audio signals in accordancewith claim 43, wherein said preselected orientation is an angleapproximately thirty degrees relative to said human head.
 45. A methodfor processing audio signals in accordance with claim 43, wherein saidpreselected orientation is a measured angle.
 46. In an audio systemhaving a plurality of directional channels first audio signal and asecond audio signal, said first and second audio signals representingadjacent directional channels on the same lateral side of a listener ina normal listening position, a method for processing said audio signals,comprising, dividing said first audio signal into a first spectral bandsignal and a second spectral band signal; scaling said first spectralband signal by a first time varying calculated scaling factor to createa first signal portion; and scaling said first spectral band signal by asecond time varying calculated scaling factor to create a second signalportion.
 47. A method for processing audio signals in accordance withclaim 46, further comprising, filtering said first signal portion by afirst filter to produce a filtered first signal portion, and filteringsaid second signal portion by a second filter to produce a filteredsecond signal portion.
 48. A method for processing audio signals inaccordance with claim 47, further comprising, combining said filteredfirst signal portion with said second audio signal to produce a firstcombined signal.
 49. A method for processing audio signals in accordancewith claim 48 with an audio system including a directional loudspeakerunit, said combining further including combining said second spectralband signal and said filtered second signal portion so that said firstcombined signal includes said filtered first signal portion, saidfiltered second signal portion, said second spectral band signal, andsaid second audio signal, said method further comprising,electroacoustically transducing, by said directional loudspeaker unit,said first combined signal.
 50. A method for processing audio signals inaccordance with claim 48 with an audio system further including adirectional loudspeaker unit and a loudspeaker unit distinct from saiddirectional loudspeaker unit, said method further comprising, combiningsaid second spectral band signal and said filtered second signal portionto produce a second combined signal; electroacoustically transducing, bysaid loudspeaker unit, said second combined signal; andelectroacoustically transducing, by said directional loudspeaker unit,said first combined signal.
 51. A method for processing audio signals inaccordance with claim 48 with an audio system further including adirectional loudspeaker unit and a loudspeaker unit distinct from saiddirectional loudspeaker unit, said distinct loudspeaker unitsubstantially limited to radiating spectral components in said firstspectral band, said combining further comprising, combining said secondspectral band signal so that said first combined signal includes saidfiltered first signal portion, said second spectral band signal, andsaid second audio signal, said method further comprising,electroacoustically transducing, by said directional loudspeaker unit,said first combined signal; and electroacoustically transducing, by saidloudspeaker unit, said filtered second signal portion.
 52. In an audiosystem having an audio signal, a first electroacoustical transducerdesigned and constructed to transduce sound waves in a frequency rangehaving a lower limit, and a second electroacoustical transducer designedand constructed to transduce sound waves in a frequency range having asecond transducer lower limit that is lower than said first transducerlower limit, a method for processing audio signals, comprising, dividingsaid audio signal into a first spectral band signal and a secondspectral band signal; scaling said first spectral band signal by a firstscaling factor to create a first portion signal; scaling said firstspectral band signal by a second scaling factor to create a secondportion signal; transmitting said first portion signal to said firstelectroacoustical transducer for transduction; and transmitting saidsecond portion signal to said second electroacoustical transducer fortransduction
 53. A method for processing audio signals in accordancewith claim 52, wherein said audio signal corresponds to a directionalchannel in a multichannel audio system.
 54. A method for processingaudio signals in accordance with claim 1, further comprising timedelaying said first spectral band signal relative to said secondspectral band signal.